ZyXEL GS-105I User Guide - Page 246
Voice Coding
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Chapter 21 VoIP The following table shows the SIP call progression. Table 105 SIP Call Progression UA 1 PROXY 1 Invite Invite 100 Trying ACK RTP 200 OK 180 Ringing 200 OK PROXY 2 Invite 100 Trying 180 Ringing 200 OK UA 2 180 Ringing 200 OK RTP BYE 1 User Agent 1 sends a SIP INVITE request to Proxy 1. This message is an invitation to User Agent 2 to participate in a SIP telephone call. Proxy 1 sends a response indicating that it is trying to complete the request. 2 Proxy 1 sends a SIP INVITE request to Proxy 2. Proxy 2 sends a response indicating that it is trying to complete the request. 3 Proxy 2 sends a SIP INVITE request to User Agent 2. 4 User Agent 2 sends a response back to Proxy 2 indicating that the phone is ringing. The response is relayed back to User Agent 1 via Proxy 1. 5 User Agent 2 sends an OK response to Proxy 2 after the call is answered. This is also relayed back to User Agent 1 via Proxy 1. 6 User Agent 1 and User Agent 2 exchange RTP packets containing voice data directly, without involving the proxies. 7 When User Agent 2 hangs up, he sends a BYE request. 8 User Agent 1 replies with an OK response confirming receipt of the BYE request, and the call is terminated. Voice Coding A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital signals back into analog voice signals. The Zyxel Device supports the following codecs. • G.711 is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal amplitudes at regular time intervals and converts them into digital samples. G.711 provides very good sound quality but requires 64 kbps of bandwidth. PX7511-B0 User's Guide 246